The invention relates generally to communication systems, and more particularly to communication systems that employ vocoders and information quality detection.
Packet switched networks for transmission of voice signals and other information have gained tremendous impetus in recent years, wherein coded speech packets are routed through public (Internet voice over IP) or private networks (intranet) out to the destination point. With conventional systems, it is assumed that speech quality improvement such as acoustic echo cancellation, noise suppression and volume level adjustment has been addressed prior to encoding speech packets at the point of packet origination by an originating unit such as an Internet appliance, portable communication device, non- portable device, or any other suitable unit. However, failure of an originating unit to provide an adequate level of acoustic level of echo cancellation, noise suppression and volume level adjustment, leads to degraded speech (i.e. unintelligible information) quality at the receiving end unit.
As known in the art, speech samples are echo canceled (due to the presence of acoustic echo from, for example, a sending unit as received by a receiving unit) noise suppressed and level adjusted (automatic gain control) followed by encoding to produce speech packets for transmission to a destination unit. In an Internet protocol (IP) system, transmitted speech packets are routed to the receiving unit. An ISP gateway routes the packets through the Internet or intranet to an ISP gateway associated with a receiving unit where the speech packets are subsequently transmitted to an end user for decoding. Throughout this packet-based communication process it is assumed by the infrastructure (e.g., ISP gateway) that speech quality related issues, such as echo cancellation, noise suppression and volume level adjustment, have been adequately addressed at the point of origination by the sending unit.
However, a problem can arise if an inadequate level of echo cancellation is performed or if noise suppression is not completely performed by the sending unit. Accordingly, it would be desirable to provide independent speech quality assessment preferably by an immediate infrastructure unit (for example, an immediate ISP gateway) associated with the transmitting unit.
Moreover, in wireless communication systems that provide mobile unit to mobile unit communication, each mobile unit (originating unit) may provide echo cancellation, noise suppression and volume level adjustment. However, vocoders within the infrastructure between the two mobile units may also provide additional coding that may include additional echo cancellation, noise suppression and volume level adjustment during a call. This is sometimes referred to as tandem vocoder operation since the mobile units provide vocoding, but the infrastructure units associated with the wireless infrastructure, such as a cellular base site controller or any other suitable network element, also performs additional vocoding. This may occur for both an uplink and downlink communications.
For example, a sending mobile unit may encode speech packets which are then decoded by a wireless network element such as a centralized base site controller (e.g., CBSC or any other suitable network element), then communicated over a land line network to a wireless network element that is serving a destination mobile unit. The network element serving the destination mobile unit then encodes the information received from the land line network and transmits it to the destination mobile where it is then decoded. There is a loss in speech quality due to the tandem vocoding, i.e., the encoding and decoding that occurs by the mobile and wireless network element at the transmission side, respectively, and the re-encoding and decoding by the wireless mobile unit at the receiving side. Typically during vocoding operation, a reduction in bit rate occurs, resulting in a loss of quality. This occurs twice when going from mobile unit to mobile unit between a land line network.
Accordingly, it is known to provide vocoder bypass to improve voice quality that otherwise would be compromised by tandem vocoding required at the point of interface to a circuit switch network. For example, vocoder bypass technology may be used for mobile to mobile telephony to protect voice quality degradation due to tandem vocoding at the point of interface to a certain circuit-switched-based network such as a PSTN. For example, Telecommunication Industry Association (TIA), Working Group 4, IS634 describes an example of vocoder bypass operation to bypass vocoding by a network element.
Referring to FIG. 1.1, mobile user 1 has to perform echo cancellation, noise suppression and rate determination followed by speech encoding to transmit speech packets to its assigned centralized base site controller (CBSC). In the absence of vocoder bypass mode, since presently CBSCs are networked via PSTN circuit switch transport, speech packets from mobile user 1 are decoded at CBSC 1 to produce 64 kilobits per second in 8 bit MU/A-law PSTN transport format and are subsequently routed to mobile user 2""s CBSC 2. At CBSC 2, the 8-bit MU/A-law speech samples are expanded into linear speech samples followed by echo cancellation, noise suppression, rate determination and encoded to produce speech packets for transmission to mobile user 2. This process is referred to as tandem vocoding which leads to degraded speech quality.
FIG. 1.2 illustrates a wireless communication network employing conventional vocoder bypass technology. As network elements, CBSC1 and/or CBSC2 employ vocoder bypass, the 16 kilobit per second rate adapted speech packets received by CBSC1 from mobile user 1 again are decoded into 64 bit per second 8 bit MU-law packets with two bits being robbed to insert rate-adapted 16 kilobit per second speech packets from mobile user 1, assuming communication from mobile user 1 to mobile user 2. Subsequently after going through the PSTN transport, at CSBC2, these two bits are extracted for transmission to mobile user 2. Thus, decoding and subsequent encoding of mobile user 1 speech packets have been bypassed at CBSC2 leading to superior speech quality compared to tandem vocoding. Using vocoder bypass, it is assumed at CBSC2 that transmitted speech packets from mobile user 1 are echo canceled, noise suppressed and volume level adjusted by mobile user 1. Hence, the CBSC2 does not monitor the speech quality of the packets. However, it has been observed that a significant number of mobile units may provide very poor acoustic echo cancellation. Thus, although vocoder bypass may improve speech quality over tandem vocoding, voice quality may suffer from the presence of uncanceled acoustic echoes. As a result, the inherent assumption with vocoder bypass mode that the underlying speech quality issues, such as echo cancellation, noise suppression and volume level adjustment, have been addressed by the transmitting mobile prior to the encoding process, may not be sufficient. Accordingly, conventional systems that employ vocoder bypass may still have echo compromised speech quality. As known in the art, echo cancellers are required to cancel echoes engendered either by a telephone network 2-4 hybrid (electrical echo) or by acoustic coupling of mouth and earpiece in a hands-free Internet appliance.
Accordingly, a need exists for independent speech quality assessment by or for a network element to improve system speech quality. In addition, it would be desirable to provide independent speech quality enhancement in systems that employ vocoder bypass or that have senders that perform echo cancellation or other speech quality enhancement.